Jonathan H
2018-Jul-28 18:06 UTC
[asterisk-users] Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?
Using pjsip 2.7.2 on Asterisk 15.5 Really struggling to make sense of translating these old 1.8 SIP instructions into a neat pjsip_wizard conf suitable for 2018 http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 In pjsip_wizard.conf, I have the following, which seems to get me registered, and it responds to an incoming call, but I always get this: [Jul 28 18:32:29] NOTICE[22492]: res_pjsip/pjsip_distributor.c:659 log_failed_request: Request 'INVITE' from '"demo" <sip:myusername at sip2sip.info>' failed for 'x.x.x.x:5060' (callid: 5fa139428fef42d9bd0cd4063e10b047) - No matching endpoint found here's what I have in pjsip_wizard.conf [sip2sip] type = wizard sends_auth = yes accepts_registrations = yes transport = simpletrans outound_auth/username = myusername at sip2sip.info outound_auth/password = password remote_hosts = 81.23.228.129,85.17.186.7,81.23.228.150,sip2sip.info endpoint/allow = alaw endpoint/context = fromsip2sip aor/max_contacts = 3 registration/contact_user = myusername outbound_proxy = proxy.sipthor.net endpoint/language=en_GB in pjsip.conf [simpletrans] type = transport protocol = UDP bind = 0.0.0.0 [acl] type = acl deny = 0.0.0.0/0.0.0.0 ; next 3 are for sip2sip permit = 81.23.228.129 permit = 85.17.186.7 permit = 81.23.228.150 in extensions.conf, I've got a bit OTT and covered every possible base to match an endpoint! Every single item in the "to" or "from" header is accounted for somewhere, so why can't it find this endpoint? Would be really grateful. Thanks. extensions.conf [fromsip2sip] exten => _.,1,Verbose(answered) [myusername] exten => _.,1,Verbose(answered) [myusername at sip2sip.info] exten => _.,1,Verbose(answered) [demo] exten => _.,1,Verbose(answered)
Joshua Colp
2018-Jul-28 20:55 UTC
[asterisk-users] Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?
On Sat, Jul 28, 2018, at 3:06 PM, Jonathan H wrote:> Using pjsip 2.7.2 on Asterisk 15.5 > Really struggling to make sense of translating these old 1.8 SIP > instructions into a neat pjsip_wizard conf suitable for 2018 > http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 > > In pjsip_wizard.conf, I have the following, which seems to get me > registered, and it responds to an incoming call, but I always get > this: > > [Jul 28 18:32:29] NOTICE[22492]: res_pjsip/pjsip_distributor.c:659 > log_failed_request: Request 'INVITE' from '"demo" > <sip:myusername at sip2sip.info>' failed for 'x.x.x.x:5060' (callid: > 5fa139428fef42d9bd0cd4063e10b047) - No matching endpoint found > > here's what I have in pjsip_wizard.conf > > [sip2sip] > type = wizard > sends_auth = yes > accepts_registrations = yes > transport = simpletrans > outound_auth/username = myusername at sip2sip.info > outound_auth/password = password > remote_hosts = 81.23.228.129,85.17.186.7,81.23.228.150,sip2sip.info > endpoint/allow = alaw > endpoint/context = fromsip2sip > aor/max_contacts = 3 > registration/contact_user = myusername > outbound_proxy = proxy.sipthor.net > endpoint/language=en_GBThis is an ITSP trunk, you've configured it kind of as if it were a phone. Instead of "accepts_registrations" you likely want "sends_registrations". Asterisk needs to register to them. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org
Jonathan H
2018-Jul-28 21:27 UTC
[asterisk-users] Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?
Thanks, but... whoah! I think I just found a bug! As soon as I changed accepts_registrations = yes to sends_registrations = yes and did a pjsip reload, Asterisk crashed. I tried starting asterisk. Nothing. In the syslog: Jul 28 22:20:41 televox kernel: [ 50.728769] asterisk[1504]: segfault at 0 ip 00007f4be3e00646 sp 00007ffc32067388 error 4 in libc-2.27.so[7f4be3d4f000+1e7000] Jul 28 22:22:02 televox kernel: [ 132.413114] asterisk[1579]: segfault at 0 ip 00007f62a9ba2646 sp 00007ffc9215d408 error 4 in libc-2.27.so[7f62a9af1000+1e7000] Took that line back out, and Asterisk started again. Shall I file a bug? On Sat, 28 Jul 2018 at 21:55, Joshua Colp <jcolp at digium.com> wrote:> > > > On Sat, Jul 28, 2018, at 3:06 PM, Jonathan H wrote: > > Using pjsip 2.7.2 on Asterisk 15.5 > > Really struggling to make sense of translating these old 1.8 SIP > > instructions into a neat pjsip_wizard conf suitable for 2018 > > http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 > > > > In pjsip_wizard.conf, I have the following, which seems to get me > > registered, and it responds to an incoming call, but I always get > > this: > > > > [Jul 28 18:32:29] NOTICE[22492]: res_pjsip/pjsip_distributor.c:659 > > log_failed_request: Request 'INVITE' from '"demo" > > <sip:myusername at sip2sip.info>' failed for 'x.x.x.x:5060' (callid: > > 5fa139428fef42d9bd0cd4063e10b047) - No matching endpoint found > > > > here's what I have in pjsip_wizard.conf > > > > [sip2sip] > > type = wizard > > sends_auth = yes > > accepts_registrations = yes > > transport = simpletrans > > outound_auth/username = myusername at sip2sip.info > > outound_auth/password = password > > remote_hosts = 81.23.228.129,85.17.186.7,81.23.228.150,sip2sip.info > > endpoint/allow = alaw > > endpoint/context = fromsip2sip > > aor/max_contacts = 3 > > registration/contact_user = myusername > > outbound_proxy = proxy.sipthor.net > > endpoint/language=en_GB > > This is an ITSP trunk, you've configured it kind of as if it were a phone. Instead of "accepts_registrations" you likely want "sends_registrations". Asterisk needs to register to them. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Reasonably Related Threads
- Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".
- SRV with pjsip on Asterisk 15.5: yes or no?
- How configure asterisk server extension.conf.
- Is set_var allowed with pjsip_wizard.conf ?
- Is set_var allowed with pjsip_wizard.conf ?